1. Field of the Invention
The present invention relates to a method for operating a hearing device by segmenting and transforming an input signal of the hearing device in a first transformation stage to form a multichannel first-stage transformation signal, subjecting a first-stage signal to multichannel processing to form a multichannel first-stage processed signal and transforming back the multichannel first-stage processed signal in the first transformation stage and assembling the resultant multichannel signal to form an output signal. The present invention also relates to a corresponding hearing device. In this case, a hearing device is understood as meaning any sound-emitting device which can be worn in or on the ear, in particular a hearing aid, a headset, earphones or the like.
Hearing aids are portable hearing devices used to support the hard-of-hearing. In order to meet the numerous individual requirements, different types of hearing aids are provided, e.g. behind-the-ear (BTE) hearing aids, hearing aids with an external earpiece (receiver in the canal [RIC]) and in-the-ear (ITE) hearing aids, for example concha hearing aids or canal hearing aids (ITE, CIC) as well. The hearing aids listed in an exemplary fashion are worn on the concha or in the auditory canal. Furthermore, bone conduction hearing aids, implantable or vibro-tactile hearing aids are also commercially available. In this case, the damaged sense of hearing is stimulated either mechanically or electrically.
In principle, the main components of hearing aids are an input transducer, an amplifier and an output transducer. In general, the input transducer is a sound receiver, e.g. a microphone, and/or an electromagnetic receiver, e.g. an induction coil. The output transducer is usually designed as an electroacoustic transducer, e.g. a miniaturized loudspeaker, or as an electromechanical transducer, e.g. a bone conduction earpiece. The amplifier is usually integrated in a signal processing unit (SPU). This basic design is illustrated in FIG. 1 using the example of a behind-the-ear hearing aid. One or more microphones 2 for recording the sound from the surroundings are installed in a hearing aid housing 1 to be worn behind the ear. A signal processing unit 3, likewise integrated in the hearing aid housing 1, processes the microphone signals and amplifies them. The output signal from the signal processing unit 3 is transmitted to a loudspeaker or earpiece 4 which emits an acoustic signal. If necessary, the sound is transmitted to the eardrum of the equipment wearer using a sound tube which is fixed in the auditory canal with an ear mold. A battery 5 likewise integrated in the hearing aid housing 1 supplies the hearing aid and, in particular, the signal processing unit 3 with energy.
Hearing aids perform, inter alia, two tasks. On the one hand, they ensure signal amplification in order to compensate for a loss of hearing and, on the other hand, noise must generally be reduced. Both tasks are tackled in the frequency domain, for which a spectral analysis/synthesis filter bank is required.
The design of the filter bank is subject to a multiplicity of underlying optimization criteria. The resultant filter bank is a compromise between time and frequency resolution, latency, computational complexity as well as cut-off frequency and stopband attenuation of the prototype low-pass filter.
A filter bank based on discrete Fourier transformation can be used for frequency analysis with a uniform resolution. A non-uniform resolution can be achieved by replacing the delay elements of the filter bank with all-pass filters, with a filter bank having a tree structure or with the use of wavelet transformation (T. Gülzow, A. Engelsberg and U. Heute, “Comparison of a discrete wavelet transformation and a non-uniform polyphase filterbank applied to spectral-subtraction speech enhancement”, Elsevier Signal Processing, pages 5-19, Vol. 64, issue 1, January 1998).
Most of these methods have either one stage or, as in the case of filter banks having a tree structure, a plurality of stages but have a long algorithmic delay and a low frequency resolution without the four optimization possibilities mentioned. See, commonly assigned patent application publications US 2009/0290736 A1, US 2009/0290737 A1, and US2009/0290734 A1, and their counterpart European publications EP 2 124 334 A1, EP 2 124 335 A2, and EP 2 124 482 A2.
The signal delay can be reduced, on the one hand, by using short synthesis windows (D. Mauler and R. Martin, “A low delay, variable resolution, perfect reconstruction spectral analysis-synthesis system for speech enhancement”, European Signal Processing Conference (EUSIPCO), pages 222-227, September 2007).
On the other hand, the resultant filter function can be transformed into the time domain and used there (P. Vary: “An adaptive filter-bank equalizer for speech enhancement”, Elsevier Signal Processing, pages 1206-1214, Vol. 86, issue 6, June 2006). The signal delay is additionally reduced by shortening the time domain filter or by conversion into a minimum-phase filter (H. W. Löllmann and P. Vary, “Low delay filter-banks for speech and audio processing”, in Eberhard Hänsler and Gerhard Schmidt: Speech and Audio Processing in Adverse Environments, Springer Berlin Heidelberg, 2008).
Filter banks are always a compromise between time and frequency resolution, signal delay and computational complexity. The compromise between time and frequency resolution is determined by the length and form of a prototype low-pass filter or prototype wavelet. Temporal extension of the prototype low-pass filter results in a lower time resolution and a higher frequency resolution. Furthermore, the temporal form of the prototype low-pass filter determines the compromise between the cut-off frequency and the stopband attenuation of a frequency response.
The compromise between time and frequency resolution or cut-off frequency and stopband attenuation, signal delay and computational complexity is made in advance and equally applies to all algorithms implemented in the hearing aid. This may be unfavorable since, for example, the amplification of individual bands in hearing aids requires high stopband attenuation in order to influence the remaining bands as little as possible by the amplification. In contrast, the stopband attenuation is less critical for noise reduction. Instead, a high frequency resolution is required in the lower frequency bands for high-quality noise reduction in order to enable noise reduction between the spectral harmonics of voiced sounds.